Systems and methods for dynamic audio processing

ABSTRACT

Systems and methods presented herein may provide for processing audio with increased loudness and dynamics. A first clock frequency associated with a digital audio file may be increased to a second clock frequency, which speeds up the playback of the audio. The digital audio file may then be converted to analog at the higher second frequency, where it is further modified in the analog domain. The analog audio is then converted back into digital audio at the second clock frequency. The clock frequency is then decreased back to the first clock frequency for playback or storage of the processed (i.e., modified or manipulated) digital audio file. Alternatively, the entire process can take place in the digital domain.

DESCRIPTION OF THE EMBODIMENTS

1. Field of the Embodiments

The embodiments relate generally to systems and methods for processingaudio, and, more specifically, to systems and methods for processingaudio for increased perceived loudness while retaining changes inperceived volume.

2. Background

Audio production can include the pre-recording processing, recording,mixing, and/or mastering of sound. These phases of audio production canall involve processing of audio, which includes the manipulation of theaudio to produce an improved digital audio file.

During audio processing, a representation of audio can be manipulated(e.g., enhanced) as either a digital or analog signal. A digital signal(i.e., digital audio) comprises a series of ones and zeros thatrepresent a sound wave (i.e., audio). An analog signal (i.e., analogaudio) comprises a continuous electrical signal that represents thesound wave. Digital manipulation (i.e., modulation) involves processingthe ones and zeros of the digital signal, such as via a processorexecuting a formula. Analog manipulation (i.e., modulation) involvespassing the analog signal through one or more physical components, sucha circuit containing resistors, capacitors, op amps, and/or a vacuumtube. Whereas an analog compressor is made up of physical components, adigital compressor can be a set of instructions executed by a processor,such as a plug-in that operates within a digital audio workstation(DAW).

Typically, the audio that needs processing is one or more digital audiofiles. For example, a user may select one or more .WAV filesrepresenting songs that need processing. While the audio processing maytake place entirely in the digital domain, the digital audio is commonlyconverted to analog audio and manipulated with analog audio componentsin most commercial audio production environments. This is the case, inpart, because of the pleasing audio qualities that analog components canadd to the audio. However, in environments where cost is a factor, someor all of the audio production process may be carried out digitallythrough the use of plugins and software, some of which may attempt tomodel the characteristics of physical analog equipment.

For example, the recording process involves recording sound in thedigital domain in the form of digital audio files. Often, someprocessing, such as pre-mixing, of these files will occur in order toadd some clarity or change the levels of the recorded audio, and todetermine whether additional takes are necessary.

Similarly, the mixing process can involve processing audio by raising orlowering levels for particular tracks, adding effects, addingequalization, adding compression, and so forth, in order to create aclearer sounding audio production.

As another example, the mastering process involves enhancing recordedaudio from a source, such as a compact disc (CD) containing a final mixof the recorded audio, to create a master version of the audio withimproved sound translation and increased loudness for the best possibleplayback on various sound systems. The enhancement almost alwaysincludes modifying the audio by applying some form of compression,limiting, and/or equalization to the audio.

The end goal of the mastering process is typically to create a masterversion of the enhanced audio that can be used to replicate and/ordistribute the audio. For example, the master audio may be storeddigitally on a compact disk. Alternatively, an analog version of themaster audio may be stored on tape or vinyl. In either case, the mediumholding the final audio is referred to as the “master,” and is generallyused to replicate the audio, such as in the creation of vinyl, compactdiscs, digital files for download, or other music media for public use.

Mastering and mixing engineers and/or home users almost always need toapply corrective equalization and/or dynamics processing (e.g.,compression and/or limiting) in order to improve upon sound translationon all playback systems and increase loudness. When processing audio,dynamics processing (e.g., dynamic compression or limiting) is used toincrease the volume of the recorded audio to two or three times theoriginal volume so that the volume level can be competitive with that ofother music in the market for sale. Achieving competitive volume levelsis important so that the mastered song is not perceived as quieterand/or less energetic than other songs played on a listener's soundsystem. However, this type of dynamic enhancement usually flattens thevolume levels and dynamic changes in the audio, removing fluctuation indynamics (loud parts vs. quiet parts) so that the listener is less ableto distinguish volume changes in the music and the impact of dynamicinstruments like drums. This type of compression and limiting is verycommon and the increases in levels can also cause audible distortion inthe music.

Similar techniques are used, for example, to ensure that commercials areloud enough to stand out and catch the attention of viewers.Additionally, mixing engineers for television and movies process sounds,voices, music, etc. in order to achieve levels and clarity that isappropriate for the particular application.

In addition to audio professionals (e.g., mastering engineers, mixers,mixers for film (television and movie audio), audio engineers, audioproducers, recording studio engineers, studio musicians, etc.), homeenthusiasts and hobbyists may also be involved with various aspects ofaudio production. For example, some people record, mix, remix, master,and/or otherwise produce audio, such as music, as a hobby. Other peopleare stereo enthusiasts (e.g., audiophiles) who use hardware and/orsoftware to process “finished” audio to achieve a better listeningexperience. Production of audio at nearly any level involves some formof audio processing. However, these hobbyists and at-home enthusiastsare often limited by their lack of training and the expense required topurchase professional-level equipment for achieving commercial-levelloudness without destroying dynamics and/or introducing distortion.

Therefore, a need exists for systems and methods of processing audiothat can achieve commercially competitive audio levels withoutdestroying the dynamics (i.e., perceived volume changes) of the song orcausing distortion in the audio.

Accordingly, systems and methods are provided herein for processingaudio to bring the volume levels up to today's very loud digital levels(or louder) while reducing distortion and retaining more volume dynamics(i.e., perceived changes in volume) than prior systems have ever allowedin the past.

SUMMARY

Embodiments described herein include systems and methods for processingaudio. In one embodiment the, system comprises a processor that plays adigital audio file. The digital audio file may contain metadataspecifying a first clock frequency for normal playback. However, theprocessor plays the digital audio file at a second clock frequency thatis higher than the first (i.e., normal) clock frequency, resulting infaster than normal playback.

In one embodiment, a digital-to-analog converter converts the digitalaudio into an analog signal (representing analog audio) while thedigital audio is playing at the higher second clock frequency (i.e.,faster than normal). This may raise the low frequency information tobecome higher frequency information of the digital audio file duringplayback, as compared to playback at the first clock frequency. Thesystem may then pass the converted analog signal through an analogcircuit to manipulate at least one sound characteristic of the analogaudio. For example, the analog circuit may contain components forcompressing, limiting, and/or making equalization adjustments to theanalog audio.

Upon passing through the analog circuit, the system may route themanipulated analog signal to an analog-to-digital converter. Theanalog-to-digital converter may then convert the manipulated analogsignal into a manipulated digital audio file, which is stored on acomputer-readable storage medium. The processor then changes the clockfrequency associated with the modified digital audio file back to thefirst (i.e., original and normal) clock frequency, for normal playback.This can lower the frequency range of the modified digital audio file tofrequencies representative of the original digital audio file (asidefrom adjustments made using, for example, equalization duringprocessing).

In one embodiment, the audio processing is carried out across multipleworkstations and/or processors. For example, a first workstation mayoutput the digital audio file to an analog circuit, which in turnoutputs to a second workstation that converts the analog audio into amodified digital audio file. This may be thought of as a “throw andcatch” arrangement.

In another embodiment, the system includes a monitoring circuit thatconverts a segment of the modified analog audio into a preview segmentof digital audio that is played back for monitoring at the first clockfrequency prior to the creation of the entire modified digital audiofile.

In another embodiment, the manipulation of the digital audio file occursentirely within the digital domain.

It is to be understood that both the foregoing general description andthe following detailed description are exemplary and explanatory onlyand are not restrictive of the embodiments, as claimed.

BRIEF DESCRIPTION OF THE DRAWINGS

The accompanying drawings, which are incorporated in and constitute apart of this disclosure, illustrate various embodiments and aspects ofthe present invention. In the drawings:

FIG. 1A is an exemplary illustration of a system for processing audio,in accordance with an embodiment;

FIG. 1B is an exemplary illustration of an alternate system forprocessing audio, in accordance with an embodiment;

FIG. 2 is an exemplary illustration of an audio processing device, inaccordance with an embodiment;

FIGS. 3A-B are exemplary flow charts with non-exhaustive listings ofsteps that may be performed in an audio processing environment, inaccordance with an embodiment;

FIG. 4 is an exemplary flow chart with a non-exhaustive listing of stepsthat may be performed by a digital audio workstation and an audioprocessing device that interface with one another, in accordance with anembodiment; and

FIG. 5 is an exemplary flow chart with a non-exhaustive listing of stepsthat may be performed by a digital audio workstation (DAW).

DESCRIPTION OF THE EMBODIMENTS

Reference will now be made in detail to the present exemplaryembodiments, including examples illustrated in the accompanyingdrawings. Wherever possible, the same reference numbers will be usedthroughout the drawings to refer to the same or like parts.

Exemplary embodiments herein allow a user to create audio files that canbe perceived as louder, more dynamic, and/or less distorted than audiofiles created using traditional methods. In one embodiment, a digitalaudio file is assigned a clock frequency (i.e., second clock frequency)that is higher than the normal playback frequency (i.e., first clockfrequency). Then the digital audio file may be played at the higherclock frequency (resulting in a faster playing speed and higherfrequency information in the audio). In one embodiment, the digitalaudio file playing at the second frequency may then be converted to ananalog signal and processed using one or more analog equalizers and/oranalog dynamics processors (e.g., compressor, limiter, etc.) (i.e., ananalog circuit). The analog audio signal (playing at the faster speed)may then be converted to digital by an analog-to-digital converter. Inthe digital domain, further processing can be applied in one embodiment.The resulting modified digital audio file may be saved to anon-transitory computer-readable medium, where the clock frequency ofthe resulting digital audio file is reset to the original normalplayback frequency so that the modified digital audio file can be playedat its original speed.

By processing the digital audio file while it plays at the higherfrequency, less low frequencies are present in the digital audio file(and more high frequency information is present), and distortionattributable to passing low frequencies through digital-to-analogconverters, compressors, limiters, equalizers, and/or other componentsmay be reduced. This results in increased capability to make the audiolouder, which in turn can result in a louder, clearer, and more dynamicaudio file. Additionally, the faster playback speed can allow for fasteraudio processing when the entire analog audio signal must be convertedback into and stored as a modified digital audio file.

Consequently, an embodiment herein may help users (e.g., masteringengineers, television or film mixers (mixing for film), home stereoenthusiasts (audiophiles), and/or anyone else who processes audio)create audio files with a competitive volume without distortion ordiminished dynamics. For example, an embodiment also may help recordingstudios create listening versions of clients' recordings at acompetitive volume so that the recording may be much louder but notdistorted. As another example, a further embodiment may allow ‘at home’studio engineers to create competitive and quality sounding recordingswithout spending money on a mastering engineer. Because an embodimentmay allow for processing audio to contain higher volume levels withminimal difficulty, this may allow hobbyists and at-home enthusiasts ameans of creating commercially-acceptable productions with reduced costsand/or training.

The methods disclosed herein may be executed in full or in part, forexample, by a processor that executes instructions stored on anon-transitory computer-readable storage medium. Similarly, a systemdescribed herein may include a processor and a memory, the memory beinga non-transitory computer-readable storage medium. As used herein, anon-transitory computer-readable storage medium refers to any type ofphysical memory on which information or data readable by a processor maybe stored. Examples include random access memory (RAM), read-only memory(ROM), volatile memory, nonvolatile memory, hard drives, solid statedrives, CD ROMs, DVDs, flash drives, disks, and any other known physicalstorage medium.

Additionally, singular terms, such as “processor,” “memory,” and“computer-readable storage medium,” may additionally refer to multiplestructures, such a plurality of processors, memories, and/orcomputer-readable storage mediums. The same applies to the termcomputer, which is understood to contain at least one processor that iscommunicatively coupled to at least one memory.

As referred to herein, a “memory” may comprise any type ofcomputer-readable storage medium unless otherwise specified. Acomputer-readable storage medium may store instructions for execution bya processor, including instructions for causing the processor to performsteps or stages consistent with an embodiment herein. Additionally, oneor more computer-readable storage mediums may be utilized inimplementing a computer-implemented method. The term “computer-readablestorage medium” should be understood to exclude carrier waves andtransitory signals.

Additionally, although “mastering” may be used as an example throughout,it is understood that the following description applies to other formsof audio production and/or audio processing, such as mixing, recording,pre-recording, and other forms of post-production.

FIG. 1A is an exemplary illustration of a system 100 for processingaudio, in accordance with an embodiment. In this example, the componentsof the system are split into the digital domain 110 and analog domain160.

In particular, the system 100 may include a computer (e.g., workstation)115 that stores a digital audio file 138. The computer may comprise oneor more computers (e.g., workstations). A workstation (e.g., digitalaudio workstation (DAW)) can comprise at least one processor and acomputer readable storage medium. In one embodiment, the workstation isa stand-alone device built specifically for handling audio production,mixing, and/or processing. For example, the workstation may have anintegrated mixer, audio sequencer, and/or effects capabilities. Inanother embodiment, the workstation can comprise a personal computerwith software being executed by a processor for the purpose of audioproduction, recording, mixing, and/or mastering.

In one embodiment, the digital audio file 138 is stored when thecomputer 115 records the digital audio file (e.g., in a recordingenvironment). In another embodiment, the computer 115 may simply importand store a previously-recorded digital audio file 138. For example, ata mastering studio, a client may bring a CD containing the digital audiofile 138, which is then accessed by computer 115. Alternatively, theclient may provide a link for downloading the digital audio file 138onto computer 115, such as by sharing a cloud-computing foldercontaining the digital audio file 138.

The digital audio file 138, as discussed herein, may include any fileformat that contains a representation of audio, such as .WAV, .AIFF,.MP3, SDII, AC3, DSD, or any number of audio file formats. For example,the digital audio file 138 shown in FIG. 1A is a .WAV file, which iscompatible with the Windows™ operating system and typically containsnon-compressed audio information (i.e., a relatively large file thatcontains all recorded audio information). However, other file types arepossible. For example, the digital audio file 138 can even include avideo file type, such as .AVI, to the extent that the video file typeincludes an audio track or portion.

The digital audio file 138 may also contain metadata that specifiescharacteristics of the digital audio file 138, such as the bit rate andthe sample rate. Other characteristics can also be identified in themetadata. For example, .WAV files contain a header that can indicatesurround sound and speaker positions, provide information regardingsample types, and supports defining custom extensions to the formatchunk.

The sample rate may indicate the number of samples per second used in adigital representation of an analog signal. The bit rate may indicatethe number of bits used to represent the level of the sample. In theory,the higher the sample rate and bit rate, the closer a discrete digitalaudio file represents the continuous analog audio signal that itemulates.

The normal playback or recording frequency (sample rate) can varybetween different digital audio files. The playback frequency is thesample rate indicated by the metadata, in an embodiment. For example,the standard sample rate (i.e., normal playback frequency) used fordigital audio files on audio compact discs (e.g., music CDs) is 44,100samples per second (44.1 kHz), with 16 bits of information per sample.Digital Video Discs (DVDs), on the other hand, contain digital audiofiles with a sample rate of 48 kHz and 24 bits of information persample.

For example, to playback a digital audio file recorded at 44.1 kHz, theplayback device will either read the metadata and automatically switchto a 44.1 kHz sample rate, or the user may have to select what samplerate the audio was recorded at, depending on the embodiment. If thewrong sample rate is selected the audio may playback at an incorrectspeed. Some systems may automatically sample rate convert the digitalaudio if the correct sample rate is not selected in the system. Thiswill resample the audio file so that it plays at the correct speed(maintaining the frequencies of the originally-recorded audio). Samplerate conversions generally can lead to fidelity loss and are avoided byaudio professionals if possible.

Additionally, music files can be recorded at a variety of differentsample rates (resulting in a variety of different normal playbackfrequencies). For example, some professional audio hardware provides theoption for sample rates of 88.2 kHz, 96 kHz, and/or 192 kHz. Even thoughstandard audio applications tend to call for digital audio files with44.1 kHz or 48 kHz sample rates, higher sample rates can be useful inaudio recording applications where effects are applied to ensure thatthe modified source information is as close to the original analogsignal (e.g., the signal generated by pressure on a microphonediaphragm) as possible.

In the case of audio with a non-commercial sample rate, the sample ratecan be converted to a standard sample rate (e.g., 44.1 kHz or 48 kHz) ata later time, such as when creating mixes or master versions of theaudio. Converting the sample rate involves re-approximating therepresented audio signal at the new sample rate, in order to preservethe frequencies and overall sound of the digital audio file. This is adifferent concept than changing the playback frequency of a digitalaudio file, which causes the digital audio file to play back faster orslower at higher or lower frequencies, respectively. Converting thesample rate instead maintains the frequency response of the audio.

In one embodiment, the digital audio file 138 may have metadataindicating a first clock frequency to use for normal playback. Forexample, the metadata may indicate a sample rate of 44.1 kHz. In oneembodiment, the sample rate may also be the clock frequency.

In another embodiment, the sample rate can be extrapolated into a clockfrequency to use for normal playback. For example, because each sampleof the digital audio file contains multiple bits-worth of information,if the system ties the clock to a particular amount of data to beprocessed, the actual clock frequency for playback may also depend onthe bit rate, which also may be indicated by metadata in the digitalaudio file 138. However, the sample rate indicated by metadata in manysystems indicates the actual clock frequency for normal playback,eliminating the need for the processor to calculate a different clockfrequency for use in playback. However, either embodiment is consideredto indicate a first clock frequency for a processor to use for normalplayback.

In one embodiment, the digital audio file is converted to an analogsignal. The processor (e.g., of the digital audio workstation 115)facilitates playback by routing information from the digital audio file138 at a specified playback clock frequency (e.g., sample rate). In oneembodiment, the information is routed to a digital-to-analog converterby the processor. The digital-to-analog converter converts the digitalsignal into an analog signal (used in analog domain 160), which isultimately supplied to speakers to produce pressure differences in theair that are perceived as sound.

In another embodiment, the processor routes the information to a digitalprocessor module (e.g., plugin) that emulates analog hardware. This canallow for additional digital effects to be applied to the digital audiofile 138 in the digital domain 110 in a way consistent with how effectsare applied in real time in an analog domain 160. However, the digitalaudio is not audible to a listener without first being converted into ananalog signal.

In one embodiment, the processor is included in computer 115 (which caninclude one or more computers). In another embodiment, the processor islocated outside computer 115, such as in an interface or module that iscommunicatively coupled to computer 115.

The processor may cause the audio file 138 to play at a second clockfrequency that is higher than the first clock frequency. For example,the processor may set the metadata of the digital audio file to indicatea second clock frequency for playback that is double the first clockfrequency. However, other combinations are possible, such as a 25percent higher clock frequency.

Using the second (i.e., higher) clock frequency for playback causes thedigital audio file 138 to playback at a faster speed than normal. As aresult, the digital file exhibits higher frequency characteristics thanwhen played at the normal playback frequency, and also completesplayback sooner. For example, by doubling the clock frequency, a digitalaudio file with audio information up to 22,500 Hz can have audioinformation up to 44,100 KHz, which is far outside the range of humanhearing.

In one embodiment, the clock frequency is chosen to substantially reduceor virtually eliminate audio frequency information below 250 Hz. Thislow frequency information often creates a “muddy” sound and may be thecause of distortion created by digital-to-analog converters and/oranalog components, such as a compressor, or digital components, such asdigital audio processors. The exact clock frequency needed to raise thelow frequency information above this threshold may vary depending on thesource audio information. For example, if an audio file has substantialaudio information at 200 Hz, a 25 percent increase in clock frequencywill move that audio information to above 250 Hz. In one embodiment, theideal clock frequency is chosen automatically by the processor, whichanalyzes the digital audio file to determine which clock frequency willmove audible levels of audio information to above 100 Hz.

In one embodiment, a digital-to-analog converter may convert the digitalaudio into an analog audio signal while the digital audio is playing atthe higher second clock frequency. Because the processor plays thedigital audio file 138 at the higher second clock frequency, less lowfrequency information is passed through the converters (relative to whenthe digital audio file 138 is played at a higher frequency), which mayreduce distortion and allow for a louder analog audio signal.Eliminating and/or reducing low frequency information lightens the loadon these components (including the analog input of analog-to-digitalconverters, in which low-frequency information can account forsignificant portions of current, causing overloading and/or distortion),which can result in a clearer analog audio signal.

The audio signal, as a result, may require less compression since it isalready louder. This, in turn, may also allow for maintaining dynamicsin volume while still achieving commercial loudness levels. This mayfurther lead to more clarity in the digital-to-analog conversion, sincelower frequencies are often the cause of the most audible distortionduring the conversion process.

The digital-to-analog converters may reside on computer 115 in oneembodiment, for example, as part of a sound card. In another embodiment,the converter(s) may be located externally to computer 115.

The clock signal used to play the digital audio 138 at the higher secondclock frequency may be generated by computer 115 (e.g., by theprocessor) in one embodiment. Alternatively, a module communicativelycoupled to the computer 115 may be responsible for generating the clocksignal in another embodiment. For example, a separate clock module maybe used to reduce an effect called jitter by having the clock modulesupply the processor with a more accurate clock signal. Other modules,such as the digital-to-analog converter module, may alternatively supplythe clock signal to the processor.

Although the sample rate is changed in metadata to reflect the secondclock frequency in one embodiment, an alternate embodiment does notalter the metadata. Instead, the DAW 115 may notify an externalconverter of the playback clock frequency to use. Or the user may selectthe clock frequency on the device supplying the clock signal. Theexternal converter may not check the metadata of the digital audio file,but instead will supply the clock at the frequency indicated by the DAWor user. In this embodiment, after the audio has been processed, theresulting modified digital audio file may already contain the correctmetadata for sample rate. However, in one embodiment, the externalconverter must be notified to change the clock frequency back to thefirst frequency for normal playback.

In one embodiment, once the digital signal (created in the digitaldomain 110 during playback) is converted to an analog signal, an analogcircuit 162 may apply at least one dynamic modification to the analogsignal. The dynamic modifications (i.e., effects) applied may include atleast one of compression, limiting, and equalization. In one embodiment,additional effects are possible, such as stereo field effects, excitereffects, tape emulation effects, etc. In order to apply these effects,the analog circuit 162 may comprise one or more hardware modules, suchas modules 165, 170, and 180. The modules may comprise any knowncombination of circuitry and analog components for applying compression,limiting, and/or equalization, depending on the dynamic effect appliedby the particular module.

Additionally, each of the modules may be connected to one another in aneffects chain in one embodiment. In an effects chain, the output fromone module can serve as an input for another module. For example, acompressor module 165 may output a modified analog signal that isreceived as an input at a limiter module 170. The output of limitermodule 170 may then be received as an input of equalization module 180.In the example shown in FIG. 1A, the output of the equalization module180 could be sent to an analog-to-digital converter so that the modifiedanalog signal may be converted back into digital audio. Additionally,although the example in FIG. 1A illustrates a signal chain whereincompression is provided first, then limiting, and then equalization,effects may also be provided in other orders. For example, equalizationmay be applied before any compression in another embodiment.

In one embodiment, multiple modules of the analog circuit 162 may bepart of a single hardware module (e.g., product) that is capable ofapplying multiple effect types.

Continuing with the example of FIG. 1A, the compressor module 165 isused to compress the dynamic range of the audio signal. This type ofcompression is distinct from data compression, in which the informationis optimized for a smaller file size. In dynamic range compression,quiet sounds can be made louder by reducing the dynamic range ofloudness and amplifying the quiet sounds.

The type of compression applied may vary between embodiments. Forexample, a peak sensing compressor may respond to an instantaneous levelof the input signal. This type of compression may provide tighter peakcontrol, but can yield very quick changes in gain reduction, which undertraditional audio processing methods can lead to audible distortion.Alternatively, an averaging compressor may be used to apply an averagingfunction (such as root mean squared (“RMS”)) on the input signal beforeits level is compared to the threshold. Some compressors may includecontrols or inputs to set a compression ratio, which typicallydetermines the reduction of signal loudness, and a gain level toincrease the loudness of the audio signal. Other controls, such asattack, release, and knee control may be provided to help shape thecompression. The attack may determine the period when the compressordecreases gain to reach the level governed by the ratio. The release maydetermine the period when the compressor is increasing gain to the levelgoverned by the ratio, or, to zero dB, once the level has fallen belowthe threshold. The length of each period may be determined by the rateof change and the required change in gain. In one embodiment, the attackand release times are adjustable by the user. In another embodiment, theattack and release times determined by the circuit design and cannot beadjusted by the user.

In an embodiment, providing an audio signal at a second (i.e., higher)clock frequency reduces the distortion caused by the compressor module165. This is because less low frequencies may be presented to thecompressor module 165 than if the signal had been created by playing theaudio at the first clock frequency. Because lower frequencies can causea bottle neck in compressors, restricting how much output can beattained before distortion occurs, providing a signal with less lowfrequency information can result in less distortion when applyingcompression.

Continuing with FIG. 1A, a limiter module 170 may receive a modifiedanalog signal from compressor module 165. Limiting, as provided by thelimiter module 170, is technically another form of compression thatincludes a very high compression ratio. For example, a compression ratiobetween 60:1 and ∞:1 may be used in limiting. The purpose of limiting isgenerally to keep the audio signal level below 0 dB, to avoid“clipping.” Audio engineers and producers typically try to avoidclipping because clipping results in a harsh and typically undesirableaudio artifact. In an alternate embodiment, limiting is not appliedbecause the converters effectively limit the audio signal when thelow-frequency information is no longer present.

With prior systems, if limiting is relied on too heavily to reduce audiolevels, overload and distortion can occur. For example, when the signalprocessed by the limiter is consistently far above 0 dB, the amount ofcompression applied by the limiter can cause distortion for similarreasons as explained above with regard to compressors. But, in oneaspect, because the analog signal is created at the higher second clockfrequency, less low frequencies may be presented and outputted to andfrom the limiter module 170 than if the signal had been created byplaying the audio at the first clock frequency. Providing a signal withless low frequencies, as accomplished in an embodiment herein, canresult in less distortion during limiting.

As shown in FIG. 1A, an equalization module 180 may apply equalizationto the audio signal. Equalization may alter the frequency response ofthe audio signal, amplifying some frequencies and/or reducing somefrequencies. This can be used, for example, to emphasize differentfrequencies across the stereo field to make particular sounds,instruments, and/or voices stand out in an audio mix. However, analogequalization hardware, particularly cheap equalization hardware commonlyfound in home studios, can introduce distortion in the low frequenciesif the audio signal is too loud for the equalizer to handle. Therefore,by using an audio signal generated according to a second (i.e., higherclock frequency), less low-end frequency information is effected by anysuch distortion.

In the example of FIG. 1A, once the modified analog signal is outputfrom the last effects module (e.g., equalization module 180), themodified analog signal is converted back into a digital audio filethrough use of an analog-to-digital converter. This conversion occurswithout changing the speed of the audio file. In other words, theconverted file initially may be set to play at the second clockfrequency.

In an alternate embodiment, the analog-to-digital converter may be setto change the playback clock frequency (e.g., sample rate) of themodified audio signal as compared to the original digital audio filewithout modifying the metadata. In this instance, the playback clockfrequency supplied (e.g., using a crystal oscillator) by theanalog-to-digital converter may be changed accordingly to cause themodified digital audio file to play at the same speed as the originaldigital audio file with the second (i.e., higher) playback frequency. Inone such embodiment, the external converter may not know the contents ofthe metadata at any point in the process. In this way, no changes to thesample rate specified in metadata occur in one embodiment.

The resulting manipulated digital audio file is stored on anon-transitory computer-readable storage medium in one embodiment. Thisnon-transitory computer-readable storage medium may be located oncomputer 115 in one embodiment, such as on a disk drive or some otherstorage medium. In another embodiment, the non-transitorycomputer-readable storage medium is located on a separate product orworkstation from computer 115.

Once the manipulated digital audio file has been stored, in one aspect,the processor sets the metadata of the manipulated digital audio toindicate the first clock frequency for normal playback speed. Thiseffectively restores the frequency response of the manipulated digitalaudio file heard when the manipulated digital audio file is played,eliminating any “chipmunk effect” caused by setting the playbackfrequency to the second (i.e., higher) frequency prior to dynamicenhancement.

The processor that sets the metadata of the manipulated digital audiocan be one or more of the processors included in computer 115 in oneembodiment. However, because the term “processor” can include aplurality of processors, including processors that are part of differentdevices and/or workstations, the processor that sets the metadata of themanipulated digital audio to indicate the first clock frequency fornormal playback speed may be located somewhere besides computer 115,such as in a different workstation or device in one embodiment.

In an alternate embodiment, one or more of the analog modules 165, 170,and/or 180 are modeled in the digital domain 110. “Modeling” may includea series of algorithms or equations that emulate the effect of hardwareused in the analog domain 160 to manipulate the analog signal. Forexample, each component of a compressor module 165 may be modeled suchthat a digital effect can be created that functions similarly to theanalog counterpart. Rather than applying a particular dynamicenhancement in the analog domain 160, the modeled digital effect isinstead applied in the digital domain 110. In this alternate embodiment,digital effects modules may be employed to emulate one or more analogmodules 165, 170, 180. For example, the digital audio file 138 may stillbe played at the second (i.e., higher) frequency, during which time thedigital effects are applied to the digital audio signal. Because thedigital signal is supplied to the emulated analog circuit at the second(i.e., higher) clock frequency, results similar to those described withrespect to the analog domain 160 may be possible.

FIG. 1B is an exemplary illustration of an alternate system 190 forprocessing audio, in accordance with an embodiment. This alternateembodiment utilizes multiple workstations 115 and 185 to carry out theaudio processing. Each workstation 115 and 185 can include its ownprocessor(s). It is understood that reference to a processor herein caninclude both a first processor of the first workstation 115 and a secondprocessor of the second workstation 185.

In the illustrated system 190, a first workstation 115 may convert theoriginal digital audio file 138 a into analog, and send the analogsignal to the analog circuit 160 for processing. It is understood thatthis conversion can utilize an external converter in one embodiment.

Then, the modified (i.e., processed) audio is sent to the secondworkstation 185. In one embodiment, this includes sending the modifiedanalog signal to the second workstation 185, where it is converted intoa modified digital audio file 138 b. In this embodiment, the modifieddigital audio file 138 b can be stored on the second workstation 185 oron some other computer-readable medium.

The other aspects the system 190 in FIG. 1B can behave similarly toembodiments described with respect to FIG. 1A.

Turning to FIG. 2, an exemplary illustration of an audio processingdevice 205 is shown in accordance with an embodiment. Audio processingdevice 205 may be a standalone product in one embodiment, that connectsto a DAW.

The audio processing device 205 may receive audio information throughinput 210 a in one embodiment. This audio information may be a digitalaudio file. The digital audio file may be a portion of some largerdigital audio file in one embodiment. For example, the audio processingdevice 205 may interface with an external digital audio workstation(DAW) in one embodiment, and receive a portion of a digital audio filefrom the DAW. In one embodiment, the DAW may be executing audiosequencing and/or editing software that allows a user to select aportion of a digital audio file for manipulation. Software executed onthe DAW, such as a plugin, may facilitate communications between the DAWand the audio processing device 205 such that the workstation may exportat least a portion of a digital audio file to the audio processingdevice 205. These communications may be received by the audio processingdevice 205 through digital input 210 a. The protocol for communicationscan vary between embodiments. The DAW is able to automate the audioprocessing from within the DAW software environment in one embodiment,sending one or more commands to the audio processing device 205 tocontrol various aspects of the mastering process.

In one embodiment, the audio processing device 205 also contains anoutput 210 b for sending the manipulated digital audio file to areceiver device. The receiver device can be the DAW, or it can be someother device that includes a non-transient computer-readable storagemedium. In one embodiment, the DAW causes the audio processing device205 to export a manipulated digital audio file back to the DAW. Once theDAW receives the manipulated digital audio file, the DAW mayautomatically integrate the manipulated digital audio file into the DAWenvironment. For example, if the DAW is used in a movie productionenvironment, an audio portion of a video file may be sent to the audioprocessing device 205, and the DAW may automatically replace the audioportion with the manipulated digital audio file received from the audioprocessing device.

In another embodiment, the manipulated digital audio file is stored on acomputer-readable storage medium contained in the audio processingdevice 205, and is manually exported later, such as by connecting theaudio processing device 205 to a DAW and browsing memory contents forthe manipulated digital audio file. In this embodiment, the contents ofthe computer-readable storage medium contained in the audio processingdevice 205 may be browsed. In one embodiment, the contents are browsedfrom the DAW. In another embodiment, display 215 is capable ofdisplaying files 250 currently stored on the audio processing device205.

In one embodiment, a single cable connects the audio processing device205 to the DAW, through a single connection that encompasses both input210 a and output 210 b. This connection may be a transceiver. Theembodiments discussed herein are not limited to a specific transferprotocol. For example, USB, Firewire, Ethernet, HDMI, SATA, and SAS arejust some of the protocols that may be implemented in variousembodiments to facilitate communication and file transfers between theaudio processing device 205 and a DAW.

In one embodiment, a first level control 212 a is provided to controlthe volume level of the audio. In one embodiment the level control 212 acontrols the level of the analog audio signal before it is routedthrough the analog circuit. In another embodiment, the level control 212a controls the level of the received digital audio file. This may allowa user to raise or lower the volume of the analog or digital audio filebefore it is manipulated by audio processing. Similarly, in anotherembodiment, an output level control 212 b is provided for adjusting thevolume of the manipulated analog audio before it is sent to ananalog-to-digital converter. In another embodiment, the output levelcontrol 212 b controls the level of a manipulated digital audio file(i.e., after audio processing), before the manipulated audio file issent back to the DAW.

In one embodiment, the audio processing device 205 may containdigital-to-analog converters for converting the digital audio file intoan analog signal before audio processing. Similarly, the audioprocessing device 205 may contain analog-to-digital converters forconverting the manipulated audio back into a manipulated digital audiofile. In addition or alternatively, the audio processing device 205 maycontrol the DAW to cause the digital-to-analog conversion andanalog-to-digital conversion to occur using the converters used by theDAW. For example, if the DAW is already equipped with and/orcommunicatively coupled to high-end converters, it may be advantageousto use those DAW converters instead of converters that may be built intothe audio processing device 205. In still another embodiment, the audioprocessing device 205 is equipped with an interface for connecting toexternal converter modules. This may allow the audio processing device205 to utilize stand-alone converters for the conversion process. Theinterface can use any protocol known in the art for communicating withD/A and A/D converters.

In the example of FIG. 2, the audio processing device 205 contains adisplay 215 for assisting the user in applying various dynamicadjustments to the audio. The display 215 can be a liquid crystaldisplay in one embodiment. In another embodiment, the display 215 can bea touch screen display. In one embodiment, the display helps the usercontrol the analog circuit for applying compression 240, equalization242, and/or limiting 244.

Additionally, in one embodiment, the audio processing device 205 alsoallows the user to specify the second clock frequency that is used forplaying the digital audio file at a faster speed during conversion intoan analog audio signal. In one embodiment, the second clock frequencymay be selected based on a multiple of the original (i.e., normal)playback clock frequency. For example, as shown in FIG. 2, the user mayselect to double the playback clock frequency, which results in doublingthe frequency response characteristics of the digital audio file, andcauses playback to occur at two times the normal playback speed. In thisembodiment, the audio processing device 205 may automatically detect thefirst (i.e., normal) clock frequency of the digital audio file. This canbe done, for example, by recognizing the file type, determining theclock frequency metadata that corresponds to that file type, and thenretrieving the first clock frequency from the metadata. For example, ifthe first clock frequency is 44.1 kHz, the second clock frequency in theexample of FIG. 2 could be 88.2 kHz. In another embodiment, the user mayenter a specific clock frequency to use as the second clock frequency.

The audio processing device 205 may automatically store the first (i.e.,normal) clock frequency of the digital audio file, so that themanipulated digital audio file can be restored to the first clockfrequency after the manipulated digital audio file is created. In thisembodiment, once the manipulated digital audio file is created (e.g.,after manipulation by the analog circuit and conversion by theanalog-to-digital converter), the audio processing device 205 may setthe clock frequency value in the metadata of the manipulated digitalaudio file to indicate the first clock frequency. Thus, when themanipulated digital audio file is played back on the DAW (or, in oneembodiment, on the audio processing device), the playback will soundnormal and not have the added “chipmunk” effect.

Additionally, in one embodiment, the audio processing device 205 maysend 246 the analog audio signal to external analog devices. Forexample, output 248 a may be used to couple the audio processing device205 with an external analog device. The analog audio signal can then besent, for example, to a compressor, limiter, and/or equalization modulethat resides external to the audio processing device 205. A return 248 bmay be provided for returning the manipulated analog signal back to theaudio processing device 205.

In still another embodiment, the audio processing device 205 may containa monitoring output for listening to the audio during the masteringprocess. In general, this allows a user to hear the effects of themastering and make adjustments to the various modules (i.e., components)of the analog circuit. In one embodiment, the user may listen to thesped-up audio during manipulated by the analog circuit.

In another embodiment, a “time warp” monitoring feature is used formonitoring the analog audio at the normal playback frequency. This mayallow a user to listen to the audio without the “chipmunk” effect, andhear how the audio will sound once the playback clock frequency is resetto the first frequency. In one embodiment, the audio processing device205 may utilize a second pair of analog-to-digital converters to createshort digital audio files (i.e., monitoring files) that representsegments of the analog audio signal being manipulated. The processor maythen set the playback frequency of the short digital audio files to thefirst clock frequency, effectively slowing the playback speed to normal.The short digital audio files may then be played in succession byconverting them back into an analog signal that is sent to monitors(e.g., speakers and/or headphones).

These short audio files may range in length in various embodiments. Inone embodiment, the short digital audio files are 5 seconds long. Theaudio processing device 205 may create these short monitoring files, forexample, by converting an even shorter segment of analog audio to adigital audio file, setting the playback frequency to the slower firstfrequency. Although this technique necessarily will cause monitoring tolag a few seconds behind any dynamics modifications applied by the user,it may still allow the user to listen to segments of audio without thechipmunk effect, so that the user does not need to complete themastering process before hearing the results of the dynamicsmodifications at the normal playback frequency.

Additionally, in one embodiment with the “time warp” monitoring feature,the user can select the length of the monitoring segments. While longerlengths may allow a more natural listening experience (i.e., lesschopped up audio segments), more time will lapse between when the usermakes a dynamic adjustment (i.e., compression, limiting, and/orequalization) and when the user can actually hear the result of theadjustment at the first playback frequency.

The analog audio segments that are converted for monitoring purposes maynot be continuous. This is because audio files play slower at the firstclock frequency than at the second clock frequency. Therefore, to ensurethat the monitoring does not lag too far behind the dynamics adjustmentsmade by the user, the time interval (e.g., 5 second) specified by theuser may be used to “catch up” the monitored files, such that at thebeginning of each time interval a new monitoring segment begins nearreal time. With this method, monitoring segments of shorter lengths,such as 1 second, may allow for monitoring near real time, but withchoppy playback since each segment begins near real time based on themanipulated audio signal, which is playing at a faster speed based onbeing created with the second clock frequency.

In one embodiment, the user may manually select whether the monitoringoutput is real time monitoring of the sped up playback, or time warpedmonitoring of the manipulated audio signal.

FIGS. 3A-B are exemplary flow charts with non-exhaustive listings ofsteps that may be performed in accordance with an embodiment. At step312 a of FIG. 3A, a DAW (e.g., computer, audio processing device, etc.)receives a digital audio file having a first clock frequency for normalplayback. The digital audio file may be received by importing the filein one embodiment. The digital audio file may also be received byrecording audio onto a computer-readable medium.

As step 312 a of FIG. 3A indicates, the digital audio file has a firstclock frequency associated with it for normal playback. Similarly, instep 312 b of FIG. 3B, a processor plays a digital audio file, whereinthe digital audio file has metadata indicating a first clock frequencyto use for normal playback. Because digital audio files can be createdusing various different sample rates, the first frequency for normalplayback may correlate to the sample and bit rate of the particulardigital audio file. Normal playback includes playing the audio file backwithout changing the pitch or the speed of the digital audio file.

At step 312 b, the processor causes the digital audio file to play at asecond clock frequency that is higher than the first clock frequency.Similarly, at step 314 a, the digital audio file is played at a secondclock frequency that is higher than the first clock frequency,increasing playback speed of the digital audio file.

In one embodiment, the steps of FIG. 3A are performed purely in thedigital domain. For example, although step 314 a may include convertingthe digital audio file into an analog signal, in one embodiment thedigital audio file is played purely in the digital domain. In step 314a, playback does not necessarily require the digital audio to be audibleto a listener. In a purely digital context, the digital audio file maybe played by processing it as if it were being converted into analog butwithout using a physical digital-to-analog converter. Instead, theprocessor may route the audio information to a plugin, which utilizesthe second clock frequency. A processor may read the digital audio fileat the second playback frequency, and apply formulas (representinganalog models) to the resulting time-based array of bits to applycompression, limiting, and/or equalization. Because accurate digitalmodels of the analog components may behave similarly to thecorresponding analog components, processing the digital audio file atthe second clock frequency as described herein may provide similarbenefits to those already outlined with regard to processing the analogsignal with the analog circuit.

The other steps of FIG. 3A may also be carried out in the digital domainin one embodiment. As discussed above, at step 316, the processor mayapply audio processing (e.g., via plugin) comprising at least one ofcompression, limiting, and equalization to the audio while utilizing(e.g., internally playing the audio file at) the second clock frequency.

Then, at step 320 a, the processor may create a modified digital audiofile based on the applied audio processing. This may include saving themanipulated digital audio file, which has modified contents, over thesource (i.e., original) digital audio file. In another embodiment, themanipulated digital audio file is saved separately from the originaldigital audio file.

Finally, at step 322 a, the processor may change the playback clockfrequency of the modified digital audio to the first clock frequency,ensuring proper playback of the manipulated digital audio file.

Unlike FIG. 3A, the steps of FIG. 3B necessarily require converting thedigital audio file for processing within the analog domain. Inparticular, step 314 b includes converting the digital audio into ananalog audio signal while the digital audio is playing at the highersecond clock frequency.

Then, step 316 b includes applying at least one of compression,limiting, and equalization to the analog audio signal. This can include,for example, passing the analog audio signal through a circuitcontaining analog hardware that modifies characteristics of the audiosignal.

At step 318, the an analog-to-digital converter may convert themanipulated analog audio signal into manipulated digital audio. Then, atstep 320 b, the processor stores the manipulated digital audio on acomputer-readable storage medium. This computer readable storage mediummay be located on a separate workstation than the workstation thatplayed the digital audio file in one embodiment.

Finally, at step 322 b, the processor may set metadata of themanipulated digital audio to indicate the first clock frequency fornormal playback speed.

FIG. 4 is an exemplary flow chart with a non-exhaustive listing of stepsthat may be performed by a digital audio workstation (DAW) and an audioprocessing device 205 while they interface with one another, inaccordance with an embodiment.

In this example, box 400 contains a non-exhaustive listing of stepsperformed by the DAW. Box 405 contains a non-exhaustive listing of stepsperformed by the audio processing device.

At step 410, audio may be recorded in the DAW and stored as a digitalaudio file.

In one embodiment, the DAW receives input from the user to perform step415, which includes exporting at least a portion of the digital audiofile to the audio processing device. For example, the user may select asingle track within the sequencing environment to export. Alternatively,the user may select just a portion of a single track to export. Stillfurther, the user may select a mixdown of an entire mix to export.

In the example of FIG. 4, the digital audio file is exported prior toincreasing the playback clock frequency of the digital audio file.However, in an alternate embodiment, the frequency is increased prior toexporting the digital audio file. In that alternate embodiment, the DAWmay output a representation of the audio at the second clock frequencyfor modification, wherein the modification includes at least one ofcompression, equalization, and limiting. This representation can beanalog in one embodiment, or digital in another embodiment.

At step 420, an input interface of the audio processing device 205accepts and stores the digital audio file. The input interface caninclude an input port, receiver circuitry, and the processor, which mayreceive data according to a protocol recognized by the DAW.

At step 425, the processor increases the playback clock frequency of thedigital audio to a higher second frequency. In this way, the audioprocessing device 205 may modify a digital audio file that normally hasa first clock frequency for playback to instead have a second clockfrequency that is higher than the first clock frequency. As mentionedabove, this step may instead occur on the DAW in one embodiment.

At step 430, a digital-to-analog converter converts the digital audiofile into analog audio while the digital audio file plays at the secondclock speed.

At step 435, the analog signal is passed through the analog modificationcircuit, which applies at least one of compression, limiting, andequalization to the analog audio signal.

Then, at step 440, the analog-to-digital converter converts the modifiedanalog audio into modified digital audio.

At step 445, that modified digital audio may be stored on acomputer-readable storage medium, after which the playback clockfrequency is changed to the first frequency. This storage medium may belocated on the audio processing device 205 in one embodiment. In anotherembodiment, the computer-readable medium is located externally to theaudio processing device.

At step 450, an output interface outputs the modified digital audio. Inone embodiment, step 450 is performed in unison with step 445.

At step 460, the DAW imports the digital audio file form the audioprocessing device.

At step 470, the DAW implements the modified (i.e., manipulated) digitalaudio, which can include adding the modified digital audio file to asequencer environment to replace or provide an alternative to theportion of the digital audio file initially exported to the audioprocessing device.

Turning to FIG. 5, an exemplary flow chart is presented with anon-exhaustive listing of steps that may be performed by a digital audioworkstation (DAW) 115.

At step 510, the DAW may modify a digital audio file having a firstclock frequency for playback to have a second clock frequency that ishigher than the first clock frequency.

At step 520, the DAW may output a representation of audio based on thesecond clock frequency for modification. For example, the representationmay be digital in one embodiment, including reading the digital audiofile at the second clock frequency. In another embodiment, therepresentation may be a representative audio signal, such as the audiosignal resulting from performing a digital-to-analog conversion of thedigital audio file by the DAW or an external converter in communicationwith the DAW.

The outputted representation may then be modified externally from theDAW. This modification may include at least one of compression,equalization, and limiting, as previously discussed herein.

At step 530, after the representation of the audio is externallymanipulated, the DAW may receive a modified representation of the audioat the second clock frequency. In one embodiment, the modifiedrepresentation is a digital audio file. In another embodiment, themodified representation is an audio signal (e.g., the conversion mayoccur on the DAW).

At step 540, the DAW may store the modified representation of the audioas a modified digital audio file. If the modified representation is ananalog signal, this step includes converting the analog signal into themodified digital audio file.

At step 550, the DAW may convert the clock frequency of the modifieddigital audio file to the first clock frequency for proper playback.

Other embodiments of the invention will be apparent to those skilled inthe art from consideration of the specification and practice of theinvention disclosed herein. It is intended that the specification andexamples be considered as exemplary only, with a true scope and spiritof the invention being indicated by the following claims.

What is claimed is:
 1. A system for processing audio, the systemcomprising: a processor that plays a digital audio file, wherein thedigital audio file is associated with a first clock frequency to use fornormal playback, wherein the processor causes the digital audio file toplay faster than normal at a second clock frequency that is higher thanthe first clock frequency; a digital-to-analog converter that convertsthe digital audio into an analog audio signal while the digital audio isplaying at the higher second clock frequency; an analog circuit thatapplies at least one of compression, limiting, and equalization to theanalog audio signal; an analog-to-digital converter that converts themanipulated analog audio signal into manipulated digital audio; and anon-transitory computer-readable storage medium that stores themanipulated digital audio, wherein the processor causes the manipulateddigital audio file to be associated with the first clock frequency fornormal playback speed.
 2. The system of claim 1, wherein the secondclock frequency is double the first clock frequency, causing the digitalaudio to play at twice the normal playback speed.
 3. The system of claim1, wherein causing the digital audio file to play at the second clockfrequency includes setting metadata of the digital audio file toindicate the second clock frequency for playback.
 4. The system of claim1, wherein the analog circuit comprises: an input level control thatmanipulates a volume level of the analog audio signal prior to the atleast one of compression, limiting, and equalization; and an outputlevel control that manipulates the volume level of the analog audiosignal subsequent to the at least one of compression, limiting, andequalization.
 5. The system of claim 1, wherein the processor comprisesa plurality of processors including a first processor on a firstcomputer and a second processor on a second computer, wherein: the firstprocessor causes the digital audio file to play at the second clockfrequency, and the second processor sets metadata of the manipulateddigital audio to the first clock frequency.
 6. The system of claim 1,further including a monitoring circuit that allows a user to hear theeffect of the modification applied by the analog circuit, wherein themonitoring circuit converts portions of the analog signal less than fiveseconds long into digital audio segments, and plays the digital audiosegments at the first clock frequency while the analog signal is passingthrough the analog circuit.
 7. A method for mastering audio, the methodincluding steps comprising: receiving a digital audio file having afirst clock frequency for the playback frequency; playing the digitalaudio file at a second clock frequency that is higher than the firstclock frequency, increasing playback speed of the digital audio file;applying audio processing comprising at least one of compression,limiting, and equalization to the audio while the audio is playing atthe second clock frequency; creating a modified digital audio file basedon the applied audio processing; and changing the playback clockfrequency of the modified digital audio to the first clock frequency. 8.The method of claim 7, further comprising converting the digital audioto an analog audio signal while the digital audio is playing at thesecond frequency, wherein the second clock frequency is twice the firstclock frequency.
 9. The method of claim 8, wherein the audio processingincludes raising the volume to a level that would cause audibledistortion when applied to a second analog audio signal generated byplaying the digital audio file at the first clock frequency.
 10. Themethod of claim 8, wherein the highest frequency containing audioinformation is increased from 22,500 Hz to 44,100 KHz.
 11. The method ofclaim 7, wherein the digital audio comprises at least a portion of asong within a digital audio workstation environment containing multipleaudio tracks, wherein the modified digital audio file replaces at leastthe portion of the single audio track within the digital audioworkstation environment.
 12. The method of claim 7, wherein monitoringis provided with playback using the first clock frequency while thedigital audio file is playing at the second clock frequency.
 13. Themethod of claim 7, wherein the second clock frequency is twice the firstclock frequency.
 14. The method of claim 7, wherein the audio processingis applied in the digital domain.
 15. An audio processing device,comprising: an input interface that accepts and stores at least aportion of a digital audio file that has a first clock frequency fornormal playback; a processor that increases the playback clock frequencyof the digital audio to a higher second frequency without converting thesample rate of the digital audio; a digital-to-analog converter thatconverts the digital audio to analog audio at the second clock speed; ananalog modification circuit that applies at least one of compression,limiting, and equalization to the analog audio; an analog-to-digitalconverter that converts the modified analog audio to modified digitalaudio; a non-transitory computer-readable storage medium that stores themodified digital audio, wherein the modified digital audio is thereafterassigned the first clock frequency for normal playback; and an outputinterface that outputs the modified digital audio.
 16. The audioprocessing device of claim 15, wherein the input interface connects to adigital audio workstation, and the input interface communicates with thedigital audio workstation in order to receive the digital audio filefrom the digital audio workstation.
 17. The audio processing device ofclaim 15, wherein the processor automatically calculates the secondfrequency based on metadata in the digital audio file regarding thefirst frequency.
 18. The audio processing device of claim 15, whereinthe second clock frequency is twice the first clock frequency.
 19. Theaudio processing device of claim 15, wherein the output interfacecommunicates with a digital audio workstation running on a computer, andoutputs the digital audio to the computer for storage.
 20. A method formastering digital audio, the method including steps comprising:modifying a digital audio file having a first clock frequency forplayback to have a second clock frequency that is higher than the firstclock frequency; outputting, for modification, a representation of audiobased on the second clock frequency, wherein the representation of audiocomprises one of the digital audio file or a representative analogsignal, wherein the modification includes at least one of compression,equalization, and limiting; receiving a modified representation of theaudio based on the second clock frequency; storing the modifiedrepresentation of the audio as a modified digital audio file; andconverting the clock frequency of the modified digital audio file to thefirst clock frequency.